similar to: ast_data, mysql, md5 hashes for passwords

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. --------------------------------------------- patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file

hello, I need help with ast_data I downloaded asterisk from cvs cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot co -r HEAD asterisk and the latest ast_data. When i run ./INSTALL.txt i get : serveur1:/opt/asterisk/ast_data# ./INSTALL patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c Hunk #1 succeeded at 27 with

Hi to everybody, I have the problem that nearly no information are displayed on the Asterisk CLI (asterisk -r). In former times (before patching Asterisk 1.0.2 with ast_data 1.0.2) it looks e.g. like this: --- snip --- -- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569 -- Accepting AUTHENTICATED call from 212.202.169.118, requested format = 1024, actual format = 1024

Hi, Has anyone got MWI to work with SIP phones while using ast_data? I have asterisk setup on one machine which is reading it config from a DB on another machine. Everything seems to work except for MWI... Any successes out there? Thanks. V. _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today - it's FREE!

Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles -------------- next part -------------- An HTML attachment was

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

Hello list, Using Asterisk 1.2.29 I use the Monitor() application. In extensions.conf I have set MONITOR_EXEC to my script (for mixing files together and convert to mp3) and I set TOUCH_MONITOR on every new channel which has to be recorded. But sometimes I'm missing the recording files. I had a look to the Asterisk-Log and saw those lines: Feb 10 15:18:57 NOTICE[16772] res_monitor.c: monitor

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/7/27, Josh Coalson <xflac@yahoo.com>: > > > > > > > But how is it possible then the FLAC encoder allows files which > have > > > a bad > > > resulting MD5 to be encoded? Is it because of the bad ram, ... > this > > > incorrect MD5 is not detected during encoding? > > >

Ooops, I missed that one, sorry. I'll see if it helps. Thx. Nick Nick (Gunnar) Bluth Linux Systems Administrator Dresdner Kleinwort Wasserstein Dresdner Bank AG Global Business Services <mailto: gunnar.bluth at DrKW.com> IT Operational Integrity Voice: +49 69 263 57913 (97000 - 57913) J?rgen-Ponto-Platz 1 Fax: +49 69 263 16994 (97000 - 16994) D-60301

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as "called number", it has to be send as "keypad facility". Anyone here with some hints? The application

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hello, I have a little problem with signalling. An E100p is connected to an Alcatel PBX, wich has an E1 to the outside. Located in Germany. zapata.conf: switchtype=euroisdn pridialplan=local prilocaldialplan=local overlapdial=yes signalling=pri_cpe .... With asterisk 1.0.2 I can call from a SIP phone to a phone connected to the Alcatel and the SIP number is correctly displayed at the caller.

Hello, I cannot get JGR installed. Here is what I have tried so far http://wiki.ubuntuusers.de/R 1. sudo -s # root werden R CMD javareconf root at gunnar-laptop:~# R CMD javareconf Java interpreter : /usr/bin/java Java version : 1.5.0_18 Java home path : /usr/lib/jvm/java-1.5.0-sun-1.5.0.18/jre Java compiler : /usr/bin/javac Java headers gen.: /usr/bin/javah Java archive

Hallo Zusammen, hatte gestern die Schnauze voll von Vista und habe mir daraufhin Ubuntu installiert. Bin also mit Ubuntu ?berhaupt nicht vertraut. Nun habe ich ohne Erfolg versucht, R zu installieren. Soweit ich der Anleitung folgen konnte habe ich es so wie auf http://wiki.ubuntuusers.de/R gemacht. Im Terminal kann ich R problemlos starten. Allerdings bei der Installation von

Hello, I have a problem with a Grandstream being behind a symmetric nat. The box which does the nat is a german "Fritz Box". This one does nat for the internal network. In the internal network is a Granstream BudgeTone 100. The nat router has a dial-up connection, so ip changes on every dial-in. |------------| |------------| |--------| |Grandstream

Hi I'm debugging why echo cancellation does not work in my app using the DUMP_ECHO_CANCEL_DATA flag. What format does the data I receive have, looks like 16bit integers just dumped but when plotting it in MATLAB I do not get a raw audio data signal I expected. Kind Regards //Gunnar Gunnar Karlsson HotSwap Stockholm AB Landsv?gen 39| SE-172 63 Sundbyberg, Sweden Mobile +46 739

Hi, I use cvs stable version. I'd like to add database support on extensions.conf. Other config files into db would be a big bonus. Looking through the wiki, there are several ways... but not sure which one is the best or widely used and supported... ast_data and res_config seem to be the best candidates. I read it many times and my impression is that ast_data seems to be better. But